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Make SIP early media work more efficiently without so many reinvites
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
97
rtp.c
97
rtp.c
@@ -733,11 +733,83 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
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return cur;
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}
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
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int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src)
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{
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struct ast_rtp *destp, *srcp=NULL; /* Audio RTP Channels */
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struct ast_rtp *vdestp, *vsrcp=NULL; /* Video RTP channels */
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struct ast_rtp_protocol *destpr, *srcpr=NULL;
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int srccodec;
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/* Lock channels */
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ast_channel_lock(dest);
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if (src) {
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while(ast_channel_trylock(src)) {
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ast_channel_unlock(dest);
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usleep(1);
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ast_channel_lock(dest);
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}
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}
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/* Find channel driver interfaces */
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destpr = get_proto(dest);
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if (src)
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srcpr = get_proto(src);
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if (!destpr) {
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if (option_debug)
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ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
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ast_channel_unlock(dest);
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if (src)
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ast_channel_unlock(src);
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return 0;
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}
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if (!srcpr) {
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if (option_debug)
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ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
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ast_channel_unlock(dest);
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if (src)
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ast_channel_unlock(src);
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return 0;
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}
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/* Get audio and video interface (if native bridge is possible) */
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destp = destpr->get_rtp_info(dest);
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vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
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if (srcpr) {
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srcp = srcpr->get_rtp_info(src);
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vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;
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}
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/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
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if (!destp) {
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/* Somebody doesn't want to play... */
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ast_channel_unlock(dest);
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if (src)
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ast_channel_unlock(src);
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return 0;
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}
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if (srcpr && srcpr->get_codec)
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srccodec = srcpr->get_codec(src);
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else
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srccodec = 0;
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/* Consider empty media as non-existant */
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if (srcp && !srcp->them.sin_addr.s_addr)
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srcp = NULL;
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/* Bridge early media */
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if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, srcp ? ast_test_flag(srcp, FLAG_NAT_ACTIVE) : 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to send early media to '%s'\n", dest->name, src ? src->name : "<unspecified>");
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ast_channel_unlock(dest);
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if (src)
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ast_channel_unlock(src);
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if (option_debug)
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ast_log(LOG_DEBUG, "Setting early media SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
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return 1;
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}
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media)
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{
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struct ast_rtp *destp, *srcp; /* Audio RTP Channels */
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struct ast_rtp *vdestp, *vsrcp; /* Video RTP channels */
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struct ast_rtp_protocol *destpr, *srcpr;
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int srccodec;
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/* Lock channels */
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ast_channel_lock(dest);
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while(ast_channel_trylock(src)) {
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@@ -780,6 +852,15 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
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ast_rtp_pt_copy(destp, srcp);
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if (vdestp && vsrcp)
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ast_rtp_pt_copy(vdestp, vsrcp);
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if (srcpr->get_codec)
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srccodec = srcpr->get_codec(src);
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else
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srccodec = 0;
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if (media) {
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/* Bridge early media */
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if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
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ast_log(LOG_WARNING, "Channel '%s' failed to send early media to '%s'\n", dest->name, src->name);
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}
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ast_channel_unlock(dest);
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ast_channel_unlock(src);
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if (option_debug)
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@@ -1086,11 +1167,17 @@ void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
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rtp->rxseqno = 0;
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}
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void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
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int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
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{
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them->sin_family = AF_INET;
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them->sin_port = rtp->them.sin_port;
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them->sin_addr = rtp->them.sin_addr;
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if ((them->sin_family != AF_INET) ||
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(them->sin_port != rtp->them.sin_port) ||
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(them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
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them->sin_family = AF_INET;
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them->sin_port = rtp->them.sin_port;
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them->sin_addr = rtp->them.sin_addr;
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return 1;
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}
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return 0;
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}
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void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
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